RTCP XR Blocks for Synchronization Delay and Offset Metrics Reporting

RTCP XR Blocks for Synchronization Delay and Offset Metrics Reporting Keio University

Graduate School of Media and Governance 5322 Endo Fujisawa Kanagawa 252-0882 Japan asaeda@wide.ad.jp

Huawei Technologies Co., Ltd.

101 Software Avenue, Yuhua District Nanjing Jiangsu 210012 China Rachel@huawei.com

Huawei Technologies Co., Ltd.

101 Software Avenue, Yuhua District Nanjing Jiangsu 210012 China sunseawq@huawei.com

This document defines an RTCP XR Report Block and associated SDP parameters that allow the reporting of synchronization delay and offset metrics for use in a range of RTP applications.

This draft defines a new block type to augment those defined in , for use in a range of RTP applications. This new block type supports reporting of Initial synchronization Delay to establish multimedia session. Information is recorded about the difference between the start of RTP sessions and the time the RTP receiver acquires all components of RTP sessions in the multimedia session . It also supports reporting of the relative Synchronization offset status of an arbitrary two streams(e.g., between audio and video streams), with the same RTCP CNAME in the multimedia session. Information is recorded about the synchronization offset time of each RTP stream relative to the reference RTP stream with the same CNAME and General Synchronization Offset of zero. The metrics belong to the class of terminal related transport level metrics defined in .

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119.

The report blocks defined in this document could be used by dedicated network monitoring applications. When joining each session in layered video sessions or the multimedia sessions, a receiver may not synchronize playout across the multimedia sessions or layered video sessions until RTCP SR packets have been received on all the components of RTP sessions. The components of RTP sessions are referred to as each RTP stream for each media type in multimedia sessions or each RTP stream at each layer in the layered video sessions. For unicast session, the delay due to negotiation of NAT pinholes, firewall holes, quality-of-service, and media security keys is contributed to such initial synchronization playout. For multicast session, such initial synchronization delay varies with the session bandwidth and the number of members, the number of senders in the session. The RTP flow Initial synchronization delay block can be used to report the initial synchronization delay of these RTP streams to be synchronized beyond the information carried in the standard RTCP packet format. In the absence of packet loss, the initial synchronization delay equals to the average time taken to receive the first RTCP packet in the RTP session with the longest RTCP reporting interval. In the presence of packet loss, the media synchronization needs to based on the in-band mapping of RTP and NTP- format timestamps [RFC6051] or wait until the reporting interval has passed, and the next RTCP SR packet is sent. In an RTP multimedia session, there can be an arbitrary two streams, with the same RTCP CNAME. The RTP Flows General Synchronization Offset block can be used to report the general Synchronization offset status of these RTP streams to be synchronized beyond the information carried in the standard RTCP packet format. In the multimedia session, the first RTP packet can be chosen as the basic packet of reference RTP stream.

This block reports Initial synchronization delay beyond the information carried in the standard RTCP packet format. Information is recorded about the the difference between the start of RTP sessions and the time the RTP receiver acquires all components of RTP sessions .

The RTP Flows Initial Synchronization Delay Report Block has the following format:

The Statistics Summary Report Block is identified by the constant <RFISD>. The constant 2, in accordance with the definition of this field in Section 3 of RFC 3611. The SSRC of the media source shall be set to the value of the SSRC identifier carried in the arbitrary RTP stream in the same multimedia session. The average delay, expressed in units of 1/65536 seconds, between the RTCP packets received on all of the components RTP sessions and the beginning of session . The value is calculated based on the information contained in RTCP SR packets or the in-band mapping of RTP and NTP- format timestamps . If there is no packet loss, the initial synchronisation delay is expected to equal to the average time taken to receive the first RTCP packet in the RTP session with the longest RTCP reporting interval.

In the RTP multimedia sessions, there can be an arbitrary number of Streams and each type of media (e.g., audio or video) is sent in a separate RTP streams. The receiver associates RTP streams to be synchronised by means of RTCP CNAME contained in the RTCP Source Description (SDES) packets . This block reports the general Synchronization offset status of these RTP streams beyond the information carried in the standard RTCP packet format. Information is recorded about the synchronization offset time of each RTP stream relative to the reference RTP stream with the same CNAME and General Synchronisation Offset of zero.

The RTP Flow General Synchronization Offset Report Block has the following format:

0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | BT=TBD | Reserved | Block length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of source | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | begin_seq | end_seq | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | General Synchronization Offset of packet begin_seq | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | General Synchronization Offset of packet begin_seq+1 mod 65536| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ ....................... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | General Synchronization Offset of packet end_seq-1 mod 65536 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The RTP Flow General Synchronization Offset Report Block is identified by the constant <RFGSO>. The constant end_seq-begin_seq-1+2, in accordance with the definition of this field in Section 3 of RFC 3611. The SSRC of the media source shall be set to the value of the SSRC identifier of the reference RTP stream to which the XR relates. The first sequence number that this block reports on. The last sequence number that this block reports on plus one. This field represents the synchronization offset time of one RTP stream in milliseconds relative to the reference RTP stream with the same CNAME and General Synchronisation Offset of zero This value is calculated based on the interarrival time between arbitray RTP packet and the reference RTP packet with the same CNAME , and timestamps of this arbitray RTP packet and the reference RTP packet with the same CNAME.

Two new parameters are defined for the two report blocks defined in this document to be used with Session Description Protocol (SDP) using the Augmented Backus-Naur Form (ABNF) . They have the following syntax within the "rtcp-xr" attribute :

rtcp-xr-attrib = "a=rtcp-xr:" [xr-format *(SP xr-format)] CRLF xr-format = RTP-flows-init-syn / RTP-flows-general-syn RTP-flows-init-syn = "RTP-flows-init-syn" ["=" max-size] RTP-flow-general-syn = "RTP-flows-general-syn" ["=" max-size] max-size = 1*DIGIT ; maximum block size in octets Refer to Section 5.1 of RFC 3611 for a detailed description and the full syntax of the "rtcp-xr" attribute.

New report block types for RTCP XR are subject to IANA registration. For general guidelines on IANA allocations for RTCP XR, refer to Section 6.2 of. This document assigns two new block type values in the RTCP XR Block Type Registry: RFISD RTP Flows Initial Synchronization Delay <RFISD> RFGSO RTP Flows General Synchronization Offset Metrics Block <RFGSO> This document also registers two new SDP parameters for the "rtcp-xr" attribute in the RTCP XR SDP Parameters Registry: "RTP-flows-init-syn" "RTP-flows-general-syn" The contact information for the registrations is: Qin Wu sunseawq@huawei.com 101 Software Avenue, Yuhua District Nanjing, Jiangsu 210012, China

The new RTCP XR report blocks proposed in this document introduces no new security considerations beyond those described in .

The authors would like to thank Bill Ver Steeg, David R Oran, Ali Begen, Colin Perkins, Roni Even, Youqing Yang, Wenxiao Yu and Yinliang Hu for their valuable comments and suggestions on this document.

RTP Payload Format for Scalable Video Coding This memo describes an RTP payload format for Scalable Video Coding (SVC) as defined in Annex G of ITU-T Recommendation H.264, which is technically identical to Amendment 3 of ISO/IEC International Standard 14496-10. The RTP payload format allows for packetization of one or more Network Abstraction Layer (NAL) units in each RTP packet payload, as well as fragmentation of a NAL unit in multiple RTP packets. Furthermore, it supports transmission of an SVC stream over a single as well as multiple RTP sessions. The payload format defines a new media subtype name "H264-SVC", but is still backwards compatible to [I-D.ietf-avt-rtp-rfc3984bis] since the base layer, when encapsulated in its own RTP stream, must use the H.264 media subtype name ("H264") and the packetization method specified in [I- D.ietf-avt-rtp-rfc3984bis]. The payload format has wide applicability in videoconferencing, Internet video streaming, and high bit-rate entertainment-quality video, among others.Table of Contents &rfc3611; &rfc2119; &rfc4566; &rfc5234; &rfc3550; Rapid Synchronisation of RTP Flows This memo outlines how RTP sessions are synchronised, and discusses how rapidly such synchronisation can occur. We show that most RTP sessions can be synchronised immediately, but that the use of video switching multipoint conference units (MCUs) or large source specific multicast (SSM) groups can greatly increase the synchronisation delay. This increase in delay can be unacceptable to some applications that use layered and/or multi-description codecs. This memo introduces three mechanisms to reduce the synchronisation delay for such sessions. First, it updates the RTP Control Protocol (RTCP) timing rules to reduce the initial synchronisation delay for SSM sessions. Second, a new feedback packet is defined for use with the Extended RTP Profile for RTCP-based Feedback (RTP/AVPF), allowing video switching MCUs to rapidly request resynchronisation. Finally, new RTP header extensions are defined to allow rapid synchronisation of late joiners, and guarantee correct timestamp based decoding order recovery for layered codecs in the presence of clock skew. Monitoring Architectures for RTP Huawei

Note to the RFC-Editor: please remove this section prior to publication as an RFC.

This document is separated from draft-wu-xrblock-rtcp-xr-quality-monitoring-01 with some editorial changes and focuses on RTP Flow Initial Synchronization Delay and RTP Flows General Synchronization Offset.

Separate Synchronization Delay and Offset Metrics Block into two independent block based on comments on the list.

The following are the major changes compared to previous version 01: Clarify which synchronization is reported in section 4 and 5. Allow calculating the synchronization delay based on RTP header extension defined in RFC6051 Explain what the components of RTP session are in section 3.

The following are the major changes compared to previous version 02: Support multiple general synchronization offset reporting. Other Editorial Changes.